Decoder for decoding a media signal and encoder for encoding secondary media data comprising metadata or control data for primary media data

ABSTRACT

An encoder for encoding secondary media data including metadata and control data for primary media data is shown, wherein the encoder is configured to encode the secondary media data using adding redundancy or bandlimiting and wherein the encoder is configured to output the encoded secondary media data as a stream of digital words. Therefore, the stream of digital words may be formed such that it is capable to resist a typical processing of a digital audio stream. Furthermore, processors for processing a digital audio stream are able to process the stream of digital words, since the stream of digital words may be designed as an audio-like or analog-like digital stream.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of copending U.S. patent applicationSer. No. 15/589,839, filed Nov. 6, 2015, which is a continuation ofInternational Application No. PCT/EP2015/075987, filed Nov. 6, 2015,which is incorporated herein by reference in its entirety, andadditionally claims priority from European Applications Nos. EP 14 192907.5, filed Nov. 12, 2014, EP 15 163 198.3, filed Apr. 10, 2015 and EP15 181 428.2, filed Aug. 18, 2015, all of which are incorporated hereinby reference in their entirety.

BACKGROUND OF THE INVENTION

The present invention relates to a decoder for decoding a media signaland an encoder for encoding secondary media data comprising metadata orcontrol data for primary media data.

In other words, the present invention shows a method and an apparatusfor distribution of control data or metadata over a digital audiochannel. An embodiment shows the convenient and reliable transmission ofcontrol data or metadata to accompany an audio signal, particularly intelevision plants, systems, or networks using standard AES3 (AES: audioengineering society) PCM (pulse code modulation) audio bitstreamsembedded in HD-SDI (high definition serial digital interface) videosignals.

In the production and transmission of music, video, and other multimediacontent, the reproduction of the content can be enhanced or made moreuseful or valuable by including metadata describing characteristics ofthe content. For example, music encoded in the MP3 format has been mademore useful by including ID3 tags in the MP3 file to provide informationabout the title or artist of the content.

In video content, it is common to include not only descriptive metadata,but data for controlling the reproduction of the content depending onthe consumer's equipment and environment. For example, televisionbroadcasts and video discs such as DVD and Blu-ray include dynamic rangecontrol data that are used to modify the loudness range of the contentand downmix gains that are used to control the conversion of a surroundsound multichannel audio signal for reproduction on a stereo device. Inthe case of dynamic range control data, gains are sent for each fewmilliseconds of content in order to compress the dynamic range of thecontent for playback in a noisy environment or where a smaller range ofloudness in the program is advantageous, by optionally multiplying thefinal audio signal by the gains.

The means of inclusion of such metadata or control data in a digitalbitstream or file for delivery to consumers is well established andspecified in audio coding standards such as ATSC A/52 (standardized inAdvanced Television Systems Committee, Inc. Audio Compression StandardA/52) or MPEG HE-AAC (standardized in ISO/IEC 14496-3 and ETSI TS 101154).

However, the transmission of metadata or control data in theprofessional or creative environment, before the content is encoded intoa final bitstream, is much less standardized. Until now this informationhas been primarily static in nature, remaining constant over theduration of the content. Although, loudness control gains are dynamic,in content production standard “encoding profiles” may be established tocontrol the generation of the gains during the final audio encodingprocess. In this manner, no dynamic metadata may be recorded ortransmitted in the content creation environment.

The development of object-oriented audio systems, where sounds in two orthree dimensions are described not by levels in traditional speakerchannels or Ambisonic components, but by spatial coordinates or otherdata describing their position and size, now involves the transmissionof dynamic metadata that changes continuously, if such sounds move overtime. Also, static objects are used to allow the creation of contentwith disparate additional audio elements, such as alternate languages,audio description for the visually impaired, or home or away teamcommentary for sporting events. Content with such static objects nolonger fits into a uniform model of channels, such as stereo or 5.1surround, which professional facilities are currently designed toaccommodate. Thus, descriptive metadata may accompany each item ofcontent during production or distribution so that the metadata may beencoded into the audio bitstreams for emission or delivery to theconsumer.

Ideally, professional content formats would simply include provisionsfor such position or descriptive metadata in their structure or schema.Indeed, new formats or extensions to existing formats, such as MDA orBWF-ADM have been developed for this purpose. However, such formats arenot understood in most cases by legacy equipment, particularly fordistribution in systems designed for live or real-time use.

In such systems, legacy standards such as AES 3, MADI, or embedded audioover SDI are common. The use of these standards is gradually beingaugmented or replaced by IP-based standards such as Ravenna, Dante, orAES 67. All of these standards or techniques are designed to transmitchannels of PCM audio and make no provisions for sending dynamic ordescriptive metadata.

One technique considered for solving this problem was to encode theaudio in a “mezzanine” format using transparent-bitrate audio coding soan appropriately formatted digital bitstream also containing staticmetadata could be included. This bitstream was then formatted such thatit could be sent as PCM coded audio data over the traditional televisionplant or professional infrastructure. A common implementation of thistechnique in the television industry is the Dolby E system, carried in aPCM AES3 audio channel according to SMPTE standard ST 337.

Dolby E allowed legacy equipment designed with four PCM audio channelsto be used for the 5.1 channels needed for surround sound, and alsoinclude provisions for transmitting the “dialnorm” or integratedloudness value of the program.

Use of the Dolby E system revealed several operational shortcomings: Oneissue was the inclusion of sample rate conversion in many devices usedto embed the PCM audio signals in the SDI infrastructure of productionor distribution facilities. Sample rate conversion or resampling of theaudio signal is commonly performed to insure correct phase and frequencysynchronization of the audio data sampling clock with that of the videosampling clock and video synchronization signals used in the facility.Such resampling has a normally inaudible effect on a PCM audio signal,but changes the PCM sample values. Thus, an audio channel used fortransmitting a Dolby E bitstream would have the bitstream corrupted byresampling. In such cases, the resampling may be disabled and othermeans used to insure synchronism of the sample clocks within thefacility.

Another issue was the delay introduced by the block-transform nature ofthe audio codec employed. The Dolby E codec used one video frame(approximately 1/30 second for interlaced ATSC video) for encoding andone video frame for decoding the signal, resulting in a two-frame delayof the audio relative to the video. This involves delaying the videosignal to maintain lip-sync, introducing additional delay in thedistribution infrastructure.

A third issue is the need to program SDI routing switchers to treatinputs carrying Dolby E bitstreams as data channels instead of audiosignals. Although Dolby E contains a “guard band” around the videosignal's vertical interval to allow routing switchers to switch toanother input without loss of the Dolby E data, many routing switchersperform a cross-fade of the audio signals during such a switch toprevent audible pops or transients in normal PCM audio signals. Thesecrossfades are of 5-20 ms in duration and corrupt the Dolby E bitstreamaround the switch point.

These operational limitations resulted in most TV facilities abandoningthe use of Dolby E in favor of a strategy of normalizing the dialnormlevel of all content upon ingest to their network, so that fixeddialnorm values and dynamic range profiles could be programmed intotheir emission audio encoders.

An alternative technique sometimes used in TV facilities is to insertmetadata information into the SDI video signal itself in the VANC dataas standardized in SMPTE standard ST 2020.

Often this is combined with carriage of the metadata using the user bitsof AES3. However, ordinary SDI embedding equipment does not support theextraction of this metadata from the AES stream for insertion into VANCbits.

An additional technique sometimes used is to encode dynamic control datawithin a PCM audio signal by inserting it into the LSBs of the audiosignal. Such a technique is described in the paper “A Variable-Bit-RateBuried-Data Channel for Compact Disc” by Oomen and has been employed inimplementations of the MPEG Surround audio coding standard. However,such buried data does not survive sample rate conversion or truncationof the LSB.

A related technique is to use extra bits such as User Bits or AuxiliarySample Bits specified in the AES3 standard as a side data channelsuitable for dynamic control data. Unfortunately, many implementationsof the AES3 standard discard this information.

A further limitation of the aforementioned techniques is they areintended for use in only in a technical transmission environment. Ifthey were routed through creative equipment, such as an audio console ordigital audio workstation, even if no operations were performed on thecontaining PCM channel, it could not be guaranteed that the data paththrough the console was bit-exact, as such equipment is not designed forsuch purposes. Even if such bit-exactness could be assured, the mereaccident of touching a control fader and thus inducing a slight gainchange in the PCM channel, would corrupt the signal.

Common to all these techniques are the limitations imposed by creationand transport equipment that is designed solely for the purpose ofcarrying PCM audio signals, without consideration for the embedding ofdigital control data.

Therefore, there is a need for an improved approach.

SUMMARY

An embodiment may have an encoder for encoding secondary media datahaving metadata or control data for primary media data, wherein theencoder is configured to encode the secondary media data to obtain astream of digital words, the encoding including transforming thesecondary media data by a digital modulation or including bandlimiting,and wherein the encoder is configured to output the encoded secondarymedia data as the stream of digital words.

Another embodiment may have a decoder for decoding a media signalincluding a received stream of digital words representing encodedsecondary media data including metadata or control data for primarymedia data; wherein the decoder is configured to recover the secondarymedia data, wherein the recovering includes applying a digitaldemodulation operation or a resampling operation to obtain recoveredsecondary media data, and wherein the decoder is configured to derive abitstream from the recovered secondary media data.

According to another embodiment, a media signal may have: a stream ofdigital words representing encoded secondary media data includingmetadata or control data for primary media data.

According to another embodiment, a method for decoding a media signalincluding a received stream of digital words representing encodedsecondary media data including metadata or control data for primarymedia data may have the steps of: recovering the secondary media data,wherein the recovering comprises applying a digital demodulationoperation or a resampling operation to obtain recovered secondary mediadata, and deriving a bitstream from the recovered secondary media data.

According to another embodiment, a method for encoding secondary mediadata including metadata or control data for primary media data may havethe steps of: encoding the secondary media data to obtain a stream ofdigital words, the encoding including transforming the secondary mediadata by a digital modulation or including bandlimiting, and outputtingthe encoded secondary media data as the stream of digital words.

Another embodiment may have a non-transitory digital storage mediumhaving a computer program stored thereon to perform the inventivemethods when said computer program is run by a computer.

According to another embodiment, a data processing system may have: aninventive encoder; and an inventive decoder.

The present invention is based on the finding that secondary media data,for example metadata carrying further information of the content of afirst media signal (e. g. payload data) or control data comprising datato control the reproduction of the content of the first media data, maybe arranged in a stream of digital words that is robust against asignificant variety of signal manipulations. Embodiments show the streamof digital words as an audio-like digital signal being able to withstandor to be robust against signal manipulation which is typical for audiosignals. The signal processing might be a transformation of the samplingfrequency, an amplification or attenuation of the signal or a DC (directcurrent) offset. The transformation of the sampling frequency may beperformed e.g. if the stream of digital words is arranged in a higherorder stream such as e.g. an AES3 PCM digital audio channel, where asampling frequency of the encoder creating the stream of digital wordsis different from a sampling frequency of a signal processor, such as anAES3 digital audio interface, creating the higher order stream.Therefore, the secondary media data can be treated as a typical audiosignal and may be therefore implemented in one of multiple audiochannels in present systems, for example in special hardware intelevision (TV) studios. A special embodiment might be an SDI videosignal containing 16 audio channels, where one audio channel is used formetadata or control data. The SDI video signal may also contain one ormore video channels. The audio channels may be PCM digital audiochannels. Therefore, the metadata or control data may be encoded as arobust analog-like digital signal instead of a standard digitalbitstream, to be robust against signal manipulation typical for PCMdigital audio channels. Present systems may be extended to comprisecontrol data or metadata by replacing current encoders and decoders withencoders and decoders described below. This replacement can be achievedby a comparably inexpensive software update. Even if the encoder anddecoder are realized in hardware, further (expensive) hardware such asbroadcast equipment can remain unchanged.

Embodiments show an encoder for encoding secondary media data comprisingmetadata or control data for primary media data. The encoder isconfigured to encode the secondary media data to obtain a stream ofdigital words, the encoding comprising transforming the secondary mediadata by a digital modulation or comprising bandlimiting. Moreover, theencoder is configured to output the encoded secondary media data as astream of digital words. Therefore, the stream of digital words may beformed such that it is able to resist a typical processing of a digitalaudio stream. Furthermore, means for processing a digital audio streamare able to process the stream of digital words, since the stream ofdigital words may be designed as an audio-like or analog-like digitalstream.

Embodiments relate to the encoding. The encoding may comprise addingredundancy by the digital modulation. The digital modulation, e.g. apulse amplitude modulation, may be so that two or more bits of thesecondary media data are transmitted per digital word of the stream ofdigital words. Moreover, the encoder may output the stream of digitalwords so that the stream of digital words is transmittable over a PCMaudio channel. Furthermore, the encoder might output a further stream ofdigital words. The further stream of digital words represents theprimary media data and the further stream is separated from the streamof digital words. The primary media data may be audio data and thesecondary media data could be metadata for the audio data or controldata for the audio data. Therefore, the encoder may be configured tooutput the stream of digital words and the further stream of digitalwords so that the further stream of digital words is transmittable overa first audio PCM channel and so that the stream of digital words istransmittable over a second audio PCM channel being different from thefirst audio PCM channel. Each of the digital words of the further streamrepresenting the primary media data might have a predefined number ofbits being greater than 8 bits and smaller than 32 bits, and whereineach of the digital words of the stream of digital words may have thepredetermined number of bits as well. The encoder may further generatethe stream of digital words so that the stream of digital wordscomprises a timing reference pattern or an amplitude reference pattern.

Further embodiments show an alignment of the secondary media data.Therefore, the encoder outputs a video stream representing a sequence ofvideo images, so that the control data or meta data of the secondarymedia data related to a certain video image are related to the certainvideo image. This is advantageous, since the sequence of video imagesmay be cut at any video image or between any of consecutive video imagesand the following video image still comprises the control data or metadata related to this video image. Furthermore, the encoder may outputthe stream of digital words as a first stream of digital wordsassociated to a first video image of the sequence of video images, andto output the stream of digital words as a second stream of digitalwords associated to a second video image of the sequence of videoimages, wherein the first and second digital words are identical to eachother. This may be advantageous, if consecutive video images compriseidentical metadata or control data, to ensure that each video imagecomprises the metadata or control data referring to the video image.

Moreover, embodiments show the encoder to output the encoded secondarymedia data as the stream of digital words as a control track and tooutput up to 15 channels of the primary media data as audio tracks,wherein the control track and the audio tracks are formed in accordancewith the AES 3 standard.

Further embodiments show the encoder being configured to generate thedigital words, the digital words having 12 to 28 bits, or wherein thedigital words are sampled at a sampling rate of between 30 kHz to 55kHz, or wherein the digital words have a dynamic range of 70 to 160 dB,or have a nominal signal level of −20 dB RMS full scale. The encoder mayuse an upper frequency for bandlimiting the secondary media data beingbetween 15 kHz to 27.5 kHz for a sampling rate between 30 kHz to 55 kHz.

Embodiments further show the encoder comprising a mapper and a streambuilder. The mapper is configured for mapping the grouped secondarymedia data comprising a first number of bits into a data word comprisinga second number of bits being greater than the first number of bits.Furthermore, the grouped secondary media data is aligned with a gap to amost significant bit or a least significant bit of the data word. Thestream builder is configured for building a stream representing theencoded secondary media data using a reference pattern and a pluralityof data words. This is advantageous, since the gap enables anamplification of the grouped secondary media data by about 6 dB (or witha factor of 2) for each bit the gap comprises to the most significantbit and an attenuation of about 6 dB (or with a factor of 0.5) for eachbit the gap comprises to the least significant bit of the data word.Therefore, it does not matter whether the amplification or attenuationis applied on purpose or accidentally, since the structure of the dataword, with the mapping of the grouped secondary media data (information)to the data word, where at both ends of the grouped secondary media datapadding is applied to obtain the data word, enables bit shifting(amplification by factor 2 for each bit shifted to the most significantbit or attenuation by factor 0.5 for each bit shifted to the leastsignificant bit). Therefore, the grouped secondary media data is notcorrupted and remains valid until the amplification or attenuation isgreater than the padding.

Embodiments further show the encoder comprising a grouper for grouping abitstream of secondary media data to form grouped secondary media data.Moreover, the encoder may comprise a reference signal generator forgenerating a reference pattern indicating a reference amplitude or apredetermined timing instant in the primary media data. The streambuilder may build a stream of digital words representing encodedsecondary media data using the reference pattern or the data word. Thereference pattern may indicate a reference amplitude or a predeterminedtiming instant in the primary media data. An analysis of the referencepattern in a decoder enables the decoder to calculate an amplificationor attenuation or a DC offset applied to the stream of digital wordsafter the stream was encoded in the encoder. Furthermore, a samplingrate of the stream of digital words may be determined from thepredetermined timing instant in the primary media data.

The stream builder may further comprise a filter to low-pass filter thedata words or the reference pattern to obtain digital words comprising alength of more than one sample of a predetermined sample rate, whereinan amplitude of the digital word is weighted according to the data wordor the reference pattern, and wherein the filter is configured to add upconsecutive digital words at instants of the predetermined sample rateto obtain the stream of digital words. Applying the filter isadvantageous, since the secondary media data is more vulnerable to aresampling than normal audio data. Therefore, the filter enables thesecondary media data to withstand applied resampling steps between theencoder and the decoder or in the decoder with respect to the encoder,and to withstand the resampling step that may be used, in the decoderperiod. Moreover, the stream of digital words may be analog and againdigital converted during resampling without considerable loss. However,resampling may not be the same as converting a digital signal to ananalog signal. Analog conversion may involve filters with impulseresponses that would smear the data, and the analog-to-digitalconversion might add quantizing noise to the signal, as well as anyanalog noise (thermal or semiconductor generated noise, hum orinterference, etc). A signal which is generated using the inventiveconcept is able to withstand a resampling and an digital-to-analogconversion.

According to further embodiments, the filter is configured to obtainzero points at instants of a predetermined sample rate of a data pulse,wherein a data pulse comprises a data word comprising grouped secondarymedia data or the reference pattern. Furthermore, the stream builder isconfigured to build the stream representing the encoded secondary mediadata using the reference pattern and a plurality of data words such thatzero points of the data pulse are aligned with a maximum of a furtherdata pulse to obtain an inter-symbol-interference-free streamrepresenting the encoded secondary media data. In other words, it isadvantageous to use a Nyquist filter, since a Nyquist-filtered signalmay be decoded in the decoder without inter-symbol-interference. Inother words, it is advantageous to use a filter satisfying the Nyquistcriterion for zero inter-symbol interference. According to embodiments,the cutoff frequency of the filter may be less than 1.5 times of asampling frequency of the primary media data.

According to an embodiment, the reference signal generator generates agrouped reference pattern comprising a first number of bits. Thereference signal generator is further configured to map the groupedreference pattern into a data word comprising a second number of bitsbeing greater than the first number of bits. Alternatively, the mappermaps a grouped reference pattern comprising a first number of bits intoa data word comprising a second number of bits being greater than thefirst number of bits. The embodiments describe options to apply theformat of the data words comprising metadata or control data to thereference pattern. Advantageously, the reference pattern obtains thesame precautions against amplification or attenuation of the mediasignal than the secondary media data. Therefore, the reference signalgenerator may provide the reference pattern in a form of the mappedsecondary media data, meaning that the reference pattern comprises afirst number of bits and is mapped into a reference pattern comprising asecond number of bits being greater than the first number of bits andcomprising the same gap to the most significant bit and the leastsignificant bit as already described in the decoder and the encoder.Alternatively, the reference signal generator outputs a referencepattern comprising a first number of bits. In accordance with thesecondary media data, the mapper maps the reference pattern with a firstnumber of bits into a data word with a second number of bits.

Embodiments further show a decoder for decoding a media signalcomprising a received stream of digital words representing encodedsecondary media data comprising metadata or control data for primarymedia data. The decoder is configured to recover the secondary mediadata using manipulating the received stream of digital words withrespect to amplitudes represented by the received digital words or usingresampling. The decoder is configured to derive a bitstream from therecovered secondary media data.

Embodiments further show the decoder comprising a reference signalgenerator, a signal manipulator, and a signal processor. The referencepattern analyzer analyzes a reference pattern of the encoded secondarymedia data, wherein the reference pattern analyzer is configured todetermine an amplitude of the reference pattern or to determine apredetermined timing instant in the primary media data. The signalmanipulator manipulates the encoded secondary media data in accordancewith the analyzed reference pattern and a computed reference pattern toobtain secondary media data. The signal processor processes the primarymedia data according to the encoded secondary media data to obtain adecoded media signal. This is advantageous, since the signal processingapplied to the media signal during the encoding enables the signalmanipulator to accurately regain the media signal from the encoded mediasignal, independent from typical signal manipulations like amplificationetc.

According to embodiments, the signal manipulator comprises a sample rateconverter configured to convert a sample rate associated with thedigital words, according to a predetermined timing instant of theprimary media data indicated in the reference pattern, to apredetermined sample rate to obtain resampled digital words. This isadvantageous, since standards for audio sampling rates may be mixedduring processing of the media data. Even a small sample rate conversionfrom e.g. 48 kHz to 48.1 kHz corrupts the secondary media data since, incontrast to audio data, there is no redundancy or dependency in thesecondary media data, which comprises metadata or control data. In otherwords, consecutive symbols of the secondary media data may vary from thehighest possible value to the lowest possible value within one sample.This results in very high frequencies due to the strong changes withinthe secondary media data.

In contrast to the secondary media data, however, audio samples aretypically band-limited, meaning that audio data changes are limited to amaximum frequency determined by the sampling frequency.

Further embodiments describe the reference pattern analyzer comprising atiming instant determiner configured to determine the predefined timinginstant of the primary media data in the reference pattern in terms ofsamples of a sample rate, an upsampler configured to upsample a rangearound the determined timing instant to determine an exact position of apredetermined timing instant, and a sampling accumulator configured todetermine an exact position of the digital words within the stream ofdigital words to obtain an actual sample rate associated to the digitalwords being different from a predetermined sample rate.

Embodiments further show the reference pattern analyzer comprising again factor calculator configured to calculate an amplification orattenuation factor according to the amplitude or the reference patternand the amplitude of the computed reference pattern and wherein thesignal manipulator comprises a multiplier configured to amplify orattenuate the data words according to the amplification or attenuationfactor to obtain gain compensated data words. This is advantageous,since an amplification or attenuation of the encoded media signal is oneof the main issues which may be caused during transfer of an encoder tothe decoder. It may be applied on purpose, for example in an equalizer,if other audio channels should be amplified or attenuated on purpose oraccidentally due to a channel with the above mentioned characteristics.

According to a further embodiment, a media signal comprising a stream ofdigital words is shown. The stream of digital words represents secondarymedia data comprising metadata and control data for primary media data.

Further embodiments show the reference pattern analyzer comprising anamplitude detector configured to determine the amplitude of thereference pattern and a further amplitude of the reference pattern. Thereference pattern analyzer may further comprise an offset compensationunit configured to calculate an offset of the encoded secondary mediadata according to a drift of the amplitude of the reference pattern andthe further amplitude of the reference pattern and wherein the secondmanipulator comprises an adder configured to add the calculated offsetof the encoded secondary media data from the encoded secondary mediadata to obtain offset compensated encoded secondary media data. Theadvantages of the embodiment are similar to those of the previousembodiment of the gain factor calculator, where an offset may be appliedto the encoded secondary media data instead of a gain, e.g. during anequalization process between the encoder and the decoder, oraccidentally from a drift caused by the transmission channel.

Embodiments further show the signal manipulator comprising a demapperconfigured to demap grouped secondary media data comprising a firstnumber of bits from the data words comprising a second number of bitsbeing greater than the first number of bits. Additionally oralternatively, the signal manipulator comprises an ungrouper configuredto ungroup grouped secondary media data comprising a first number ofbits to obtain a decoded media data bitstream. The digital words mayfurther comprise the digital words comprising filtered secondary mediadata comprising a reference pattern and a plurality of data words,wherein the secondary media data is mapped into data words with a gap tothe most significant bit of the data word or the least significant bitof the data word. Moreover, the reference pattern may comprise areference amplitude of the encoded secondary media data and apredetermined timing instant in primary media data and wherein theplurality of data words comprise secondary media data.

Embodiments show the media signal comprising a further stream of theprimary media data, wherein the primary media data comprises audio dataor video data, wherein the further stream comprising primary media datais aligned to the stream of encoded secondary media data and thepredetermined timing instant in the primary media data. This isadvantageous, since the timing instant in the primary media data allowsan accurate alignment of the secondary media data to the primary mediadata. In other words, an audio signal and metadata or control data maybe aligned to frames of a video signal at a vertical blanking or afurther synchronization signal of the video signal. Furthermore, thetiming instant may be a synchronization signal in an audio signal, wherethe secondary media data is aligned to. Therefore, the secondary mediadata may be also applied to audio-only streams. The idea is to provideany information of the secondary media data within each frame of thevideo signal. Since the secondary media data is aligned to the timeinstant in the primary media data where the video stream is cut, thesecondary media data remains unchanged and is intact. Therefore, eachvideo frame may contain any information from the secondary media dataeven if the video signal comprising the video frame is cut.

Embodiments may be developed according to the following considerations.Therefore, it is an advantage of embodiments of the invention to providea means for carrying static and dynamic control data or metadataaccompanying PCM (pulse code modulation) digital audio signals throughtraditional creative and distribution equipment which only provides PCMaudio channels.

This may be accomplished by considering the PCM digital audio channel'sfundamental nature as a transmission means for an audio signal. Suchaudio signals are normally digitized for television use at a bit-depthof 16 to 24 bits and at a sampling rate of 48 kHz and have a resultingdynamic range of 90 to 140 dB, with a nominal signal level of −20 dB RMS(root mean squared) full scale.

Thus, if one considers the typical AES3 transmission channel as adigitized communication channel having these characteristics, themodulation techniques commonly employed in digital communications may beused to send modulated data over the channel. Such techniques arenaturally immune to gain changes, moderate time base distortions, and inmany cases, frequency response distortions of the channel.

The AES3 PCM digital audio channel differs from the channels used fordigital communication. It is strictly a digital channel, and does notsuffer from the multipath and rapid channel fading typical of radiocommunications channels. Given the 90 to 140 dB dynamic range, it is notpractically limited in potential transmit power to provide sufficientcarrier to noise ratio. When used in video systems, such as embedded inthe SDI (serial digital interface) video signal, it has an inherentblock nature due to the need to avoid the video vertical sync intervalwhere switching can occur. Also, unlike many communications systems,there is a need for low latency, to avoid lip-sync issues or to avoiddifficulties in monitoring audio when producing live broadcasts.

The throughput requirements of the control data or metadata needed forobject audio vary by the number of objects, whether they are static ordynamic, and the particular object audio standard employed. One suchstandard is the MPEG-H Audio specification, ISO/IEC 23008-3.

In this standard, typical use cases involve metadata or control databeing encoded in streaming packets using the MHAS (MPEG-H Audio Stream(defined in ISO/IEC 23008-3 in Chapter 14 “MPEG-H 3D audio stream))specification at bitrates of 10-30 kb/s.

For example, each dynamic object in a MPEG-H audio scene may use 1.5kb/s for transmission. Thus, a program with 16 dynamic objects (apractical maximum given that the SDI interface only supports 16 channelsof embedded audio) may use about 25 kb/s of data. Static metadataregarding the audio scene could take another 40-50 kb/s, if it was senteach audio frame.

The potential bit error rate (BER) needed can be estimated byconsidering the following factors: If a single bit error were permittedin operation once per year, given a bitrate of 75 kb/s, 2.36E12 bitswould be sent in a year, requiring a bit error rate of 4.2E-13. However,the information in the control data is highly redundant. In most cases,bit errors will be detected by the underlying MHAS protocol and thecontrol data would be interpolated from surrounding packets.Additionally or alternatively, CRC (cyclic redundancy check) values,e.g. using 16 bit, or other suitable codes or mechanisms to check forbit errors may be used. In this case, a bit error once per hour might bea reasonable upper limit. This latter case would use a BER of 3.7E-9.Thus, a reasonable BER for this transmission scheme would likely need aBER between 1 E-9 and 1E-12, which is easily possible with the highsignal to noise ratios available in the AES3 digital audio channel.

It should be noted that the typical expressions for BER forcommunications channels do not apply here, as the noise in this channelis strictly that of quantization and resampling, with a rectangular orpossibly (in the case dither is applied) triangular probability densityfunction.

The time-base error introduced by sample rate conversion (or moreprecisely, by sources operating asynchronously) is limited by theaccuracy of the clock sources employed in each piece of equipment actingas an asynchronous source. Most professional television facilitiesoperate with clock or synchronization signal sources generated fromaccurate crystal, GPS, or rubidium standards, typically with a maximumfrequency tolerance of 0.1 to 1.0 ppm. Typical consumer equipment mayhave frequency tolerances of 30 ppm. Allowing some margin for the caseof consumer equipment operating at temperature extremes, a tolerance of100 ppm may be safely assumed, for the case of consumer equipmentoperated in the field being connected to a professional TV plant.

Thus, a possible set of design assumptions and goals for applying thisinvention for the purpose of transmitting the control data or metadataneeded for a common use of the MPEG-H Audio standard are:

Sampling Frequency 48 kHz Symbol Frequency 16 kbaud (⅓ sample rate forconvenience) Desired bitrate 75 kb/s Maximum latency, end to end 240samples or 5 ms Maximum time-base error 100 ppm Channel Bit Depth 14bits (allowing for poor rounding, extra quantizing noise in poor digitalaudioequipment design, etc.) Channel Gain +15 to −20 dB (to allow forgain errors in equipment, or inadvertent adjustment of a channel gain inprocessing equipment or an audio console or workstation) Nominal RMS orloudness value −30 to −15 dB FS (to allow of signal operationa personnelto monitor the signal level of the audio channel as they would fornormal audio signals)

A further goal of a embodiment of this invention is to allow ease ofimplementation and debugging by audio coding engineers, who are veryfamiliar with the building blocks used in perceptual audio coding, butwho may not have experience with the implementation techniques common todata communications.

Given the channel bandwidth of 24 kHz, and design symbol rate of 16kbaud, simple classical modulation techniques such as ASK or PSK willnot be adequate. Modulation that provides coding efficiency of at least5 b/s/Hz will be used.

Those skilled in the art will realize that a number of commonly usedmodulation techniques for digital communications would satisfy thesedesign assumptions and goals. For example, 64 QAM (Quadrature AmplitudeModulation with an alphabet of 64 symbols) could be used, as it providesa coding efficiency of 6 b/s/Hz. However, implementing a QAM demodulatorgenerally uses moderately complex signal processing to recover thecarrier frequency and symbol clock, including the use of digital phaselock loops (PLL) which are unfamiliar to audio coding engineers. SuchPLLs involve tuning of loop filters or accumulators to avoid loopinstability, and may need some time to stably acquire the signal after atransient or switch.

The embodiment presented here uses 32 PAM (Pulse Amplitude Modulationwith 32 levels) as an alternative that does not involve PLLs andproduces a design that uses signal processing functions commonlyemployed in audio coding. PAM involves a 6 dB increase in signal tonoise ratio for each increment of coding efficiency compare to the 3 dBneeded with QAM, but in this system the signal to noise ratio isinherently high, while the design and debugging costs of a PAM receiverare lower.

All of the previously described embodiments may be seen in total or incombination, for example in a television plant where the encoder encodesa video signal with a corresponding audio signal and metadata or controldata (secondary media data), for example at a first sampling frequencyand wherein the decoder may be applied to a control instance (e. g.monitoring unit) or an emission instance before transmission of themedia signal to a consumer.

BRIEF DESCRIPTION OF THE DRAWINGS

Embodiments of the present invention will be discussed subsequentlyreferring to the enclosed drawings, in which:

FIG. 1 shows a schematic block diagram of a system of an encoder and adecoder in a television plant or a network according to embodiments;

FIG. 2 shows a schematic block diagram of an encoder for encodingsecondary media data according to an embodiment;

FIG. 3 shows a schematic block diagram of an encoder for encodingsecondary media data according to a further embodiment;

FIG. 4 shows a schematic conceptual block diagram of the transmitterportion of the invention that accepts a metadata or control databitstream and encodes it as a 32 PAM signal formatted for transmissionin a 16 bit, 48 kHz PCM audio channel according to an embodiment;

FIG. 5 shows a schematic block diagram of a decoder according to anembodiment;

FIG. 6 shows a schematic block diagram of a decoder according to anembodiment;

FIG. 7 shows a schematic conceptual block diagram of a receiver portionaccording to embodiments that accepts PCM data from a 16 bit, 48 kHz PCMaudio channel and decodes the embedded 32 PAM signal into a metadata orcontrol data bitstream according to an embodiment;

FIG. 8a shows a schematic representation of a media signal according toan embodiment;

FIG. 8b shows a schematic representation of a media signal according toa further embodiment;

FIG. 8c shows a schematic diagram showing the mapping of the 5-bit 32PAM signal symbols into the 16-bit PCM audio channel sample wordaccording to an embodiment;

FIG. 9a,b shows a schematic waveform diagram showing the timingrelationship between the video facility's vertical sync signal and theencoded metadata or control data in the PCM audio channel according toan embodiment;

FIG. 10a shows a raised cosine shape filter with a rolloff factor of0.98 in a time-continuous representation;

FIG. 10b shows a raised cosine shape filter with a rolloff factor of0.98 in a time-discrete representation;

FIG. 11a shows the raised cosine shape filter function with a rollofffactor of 0.7 in a time-continuous representation;

FIG. 11b shows the raised cosine shape filter function with a rollofffactor of 0.7 in a time-discrete representation;

FIG. 11c shows the image of FIG. 11b three times in a row, aligned withan offset of two samples between adjacent filter functions;

FIG. 12a shows a schematic representation of a stream according to anembodiment in a time-continuous representation according to anembodiment;

FIG. 12b shows a part of the stream already presented in FIG. 12a in anenlarged version;

FIG. 12c shows a schematic representation of the stream according to anembodiment in a time-discrete representation according to an embodiment;

FIG. 12d shows a part of the stream already presented in FIG. 12a in anenlarged version;

FIG. 13 shows a schematic flow diagram of a method for decoding a mediasignal comprising a stream representing secondary media data using areference pattern and a plurality of data words;

FIG. 14 shows a schematic flow diagram of a method for encoding a mediasignal with an encoder;

FIG. 15a -1 and FIG. 15a -2 show a schematic representation of a systemin a fixed mode; and

FIG. 15b -1 and FIG. 15b -2 show a schematic representation of thesystem in a Control Track Mode.

DETAILED DESCRIPTION OF THE INVENTION

In the following, embodiments of the invention will be described infurther detail. Elements shown in the respective figures having the sameor a similar functionality will have associated therewith the samereference signs.

Embodiments provide convenient and reliable transport of audio signalmetadata or control data accompanying a digital audio signal. Themetadata or control data is digitally modulated or encoded into a signaltolerant of typical transmission degradations for distribution inprofessional audio or video production or distribution facilities andnetworks over a normal digital audio channel, or the channel is embeddedin a digital video signal. Metadata may comprise a description foron-screen displays, position of objects within a video frame, languageinformation for different audio channels such as e.g. German, English,French etc. language. Control data may comprise information regarding acoding of the audio channels in order to apply the correct decodingparameters or control data may comprise parameters to interpret higherorder ambisonics or any other information to decode the audio signal.However, metadata and control data may be used for many other purposes.In digital media, essence is the underlying content of an asset, andmetadata is descriptive data about that asset. Therefore, the abovementioned examples do not limit the scope of the invention.

FIG. 1 shows a schematic block diagram of a data processing system 400comprising an encoder and a decoder. Specifically, FIG. 1 shows a 32 PAMmodulator 3 comprising the encoder and a 32 PAM demodulator 9 comprisingthe decoder. Furthermore, a media signal 155 comprising a bitstream ofsecondary media data 125 and primary media data 90 a (e.g. audio essencesignals) and additionally, the primary media data 90 b (e.g. a videosignal) are shown according to an embodiment. The system may be part ofa TV studio where the secondary media data comprising audio control dataor a metadata bitstream is included in the audio essence signals andtherefore aligned to the video signal for each video frame. Therefore,in the TV studio, the encoded video signal may be checked using amonitoring unit and therefore using the decoder to decode the encodedmedia signal. Furthermore, the media signal may be decoded using thesecondary media data before channel coding and further processingoperations to prepare the final media signal to be transmitted to aconsumer. This final media signal does not have the secondary mediasignal anymore.

More generalized, according to an embodiment, the data processing systemcomprises a signal manipulator for manipulating the stream of digitalwords to obtain a manipulated stream of digital words, wherein thedecoder is configured to recover the stream of digital words from themanipulated stream of digital words. The signal manipulator maymanipulate by amplitude amplification or amplitude attenuation or offsetintroduction or offset variation or frequency selective attenuation oramplification or resampling. Furthermore, the decoder can recover thestream of digital words manipulated by amplitude amplification oramplitude attenuation or offset introduction or offset variation orfrequency selective attenuation or amplification or resampling.Moreover, the signal manipulator can receive a PCM audio channel and mayoutput a PCM audio channel, wherein the encoder is configured to outputa signal transmittable over the PCM audio channel, and wherein thedecoder is configured to receive the transmitted stream from the PCMaudio channel.

In other words, FIG. 1 shows the operation of an embodiment of theinvention in the environment of a professional audio or video productionor distribution facility or network. Audio peripheral or workstation 1is a source of one or more digital audio signals, referred to as essencesignals (or primary media data) to distinguish them from related controldata or metadata signals (secondary media data), which are also sourcedby the peripheral or workstation.

The control data or metadata bitstream is input to transmitter 3 whichconverts it to a form such as 32 PAM modulated PCM samples which willsurvive normal channel impairments of AES3 or HD-SDI channels. Thesamples, as well as one or more optional audio essence signals are thensupplied to AES3 digital audio interface 4. The output of this interfaceis embedded in a HD-SDI video signal by embedder 5, which may applysample rate conversion to align the phase and frequency of the AES3clock with the clock and sync signals of the HD-SDI video signal. Thisvideo signal is then distributed through an SDI-based television plantor infrastructure 6 for delivery to a second audio peripheral orworkstation 2. The digital audio signals are extracted from the HD-SDIsignal by de-embedder 7 and sent as AES3 bitstreams to AES3 digitalaudio interface 8. The PCM data corresponding to the AES channelcontaining the control data or metadata information (encoded secondarymedia data) is sent to a receiver 9. The receiver 9 comprises thedecoder 50, which decodes the 32 PAM or similar modulated signals intothe audio control data or metadata bitstream 85′, which may be part ofthe decoded media signal 85. Furthermore, the signal processor 70 shownin FIG. 6 processes the primary media data (audio essence signal)according to the encoded secondary media data to obtain the encodedmedia signal.

FIG. 2 shows a schematic block diagram of an encoder 100 for encodingsecondary media data comprising metadata and control data for primarymedia data. The encoder is configured to encode the secondary media data80 using adding redundancy or bandlimiting. The encoder is furtherconfigured to output the encoded secondary media data as a stream 145 ofdigital words. In an embodiment, redundancy may be added to thesecondary media data by zero padding or sign-extension. Otherembodiments may use checksums or redundancy codes. A further embodimentshows a bandlimited secondary media data or a bandlimited group ofsecondary media data optionally with or without added redundancy.Bandlimiting may be derived by applying a (low-pass) filter to a signalor more specific, to an outbound signal of the encoder, which may be agrouped or mapped secondary media data. According to furtherembodiments, the encoder is configured to generate the digital words,the digital words having 12 to 28 bits, or wherein the digital words aresampled at a sampling rate of between 30 kHz to 55 kHz, or wherein thedigital words have a dynamic range of 70 to 160 dB, or have a nominalsignal level of −20 dB RMS (root mean square) full scale. The encodermay be also configured to use an upper frequency for bandlimiting thesecondary media data being between 15 kHz to 27.5 kHz for a samplingrate between 30 kHz to 55 kHz.

FIG. 3 shows a schematic block diagram of an encoder 100 for encoding amedia signal. The encoder 100 comprises a mapper 115, and a streambuilder 120. The mapper 115 is configured to map a group of groupedsecondary media data 130 comprising a first number of bits into a dataword 140 comprising a second number of bits being greater than the firstnumber of bits. The grouped secondary media data is aligned with a gapto a most significant bit or a least significant bit of the data word.The stream builder is configured to build a stream of digital wordsrepresenting encoded secondary media data. According to furtherembodiments, the encoder comprises a grouper 105 configured for groupingthe secondary media data 80, which may be a bitstream of secondary mediadata, to form grouped secondary media data 130. Moreover, the encodermay comprise a reference signal generator 17 configured to generate areference pattern indicating a reference amplitude or a predeterminedtiming instant in the primary media data, wherein a stream builder 120is configured to build a stream 145 of digital words representingencoded secondary media data 55 using the reference pattern 60 or thedata word 140. Therefore, both signals, the reference pattern 135 andthe data word 140 may be input to a stream builder 120 configured tobuild a stream 145 of digital words representing encoded secondary mediadata.

FIG. 4 shows a schematic block diagram of the encoder 100 according toan embodiment. Embodiments show the encoder 100 comprising a filter 15to low-pass filter the data word or the reference pattern to obtain adata pulse comprising a length of more than one sample of apredetermined sample rate, wherein the amplitude of the data pulse isweighted according to the data word or the reference pattern, andwherein the filter is configured to add up consecutive data pulses atinstants of the sample rate. Furthermore, the filter may be configuredto obtain zero points at samples of a predetermined sample rate of thedata pulse. The data pulse comprises a data word comprising groupedsecondary media data or the reference pattern. The stream builder isconfigured to build the stream representing the encoded secondary mediadata using the reference pattern and a plurality of data words such thatzero points of the data pulse are aligned with a maximum of a furtherdata pulse to obtain an inter-symbol-interference (ISI)-free streamrepresenting the encoded secondary media data. In other words, it isadvantageous to use a Nyquist filter enabling the decoder to resamplethe data words or the stream of digital words withoutinter-symbol-interference or aliasing problems. FIG. 11c shows anembodiment illustrating a filtered data word and building an exemplarystream from three of the data words. According to embodiments, thefilter comprises a cut of frequency of less than 1.5 times of a samplingfrequency of the primary media data.

It has to be noted that the mapper 115 is not depicted in FIG. 4.However, the mapper may be implemented between register 14 and thefilter 15 or being part of one of the blocks or functions.

In other words, FIG. 4 shows the conceptual operation of a transmitterportion of an embodiment of the invention. The audio control data ormetadata bitstream is input to a buffer 10 for temporary storage toallow for interruptions in the transmitted data during a vertical sync160 or other processing operations. The bitstream 125 is parallelizedinto words of 5 bits and transferred out of the buffer by conceptualregister 11. The output of the register is then encoded into a Gray codevalue by an encoder 12. Except when the vertical sync signal 160 isactive, the output of the encoder 12 is input to the register 14. Theoutput of the register 14 is taken as a two's complement binary number,which is sign-extended and mapped into a 16-bit data word as shown inFIG. 8c , and fed into a pulse shaping filter 15. The filter is ideallya Nyquist type filter that exhibits sin(x)/x nulls in its impulseresponse at symbol periods to prevent inter-symbol-interference. Suchfilters are well known in digital communications theory. For example, asuitable filter would be a raised-cosine pulse shaping filter with anexcess bandwidth parameter set to 0.75. The output of the filter 15 isthen fed to further transmission means for inclusion as audio samples inthe PCM audio channel and embedding in an SDI video signal. Theprocessing may be driven by a (PCM) sample clock 99 of e.g. 48 kHz.

During the vertical sync interval of the video signal, a conceptualswitch 13 selects the output of the reference signal generator 17 fortransmission instead of the output of a Gray encoder 12. No data is readfrom a buffer 10 during this interval. The reference signal generator 17outputs a symbol value of zero and thus a steady-state PCM value of zeroduring the vertical blanking interval. At the end of the verticalblanking interval, the reference signal generator outputs eight symbolswith code 0x0F and then eight symbols with code 0x11, before the switch13 returns to the output of Gray encoder 12 and data begins being readfrom the buffer 10. In this manner (e.g. using scaling in the filter 15)the sixteen-bit signed two's complement PCM signal shown in FIG. 4 isproduced, having a value of zero during vertical blanking, then aneight-symbol wide positive pulse 41 of value 0x0780 and then aneight-symbol wide negative pulse 42 of value 0xf880. The pulses 41 and42 thus form a positive and negative amplitude reference and a strongtransition at a symbol edge that may be used in a receiver to recoverthe original amplitude and phase of the transmitted 32 PAM signal.

FIG. 5 shows a schematic block diagram of a decoder 50 for decoding amedia signal 155 comprising a received stream 145 of digital wordsrepresenting encoded secondary media data 55 comprising metadata andcontrol data for primary media data. The decoder 50 is configured torecover the secondary media data using manipulating the received streamof digital words with respect to amplitudes represented by the receiveddigital words or using resampling and wherein the decoder is furtherconfigured to derive a bitstream 125′ from the recovered secondary mediadata. The decoder may know original amplitudes or a predetermined timinginstant of the digital words before transmission to manipulate thereceived digital words to recover the secondary media data.

FIG. 6 shows a schematic block diagram of a decoder 50 for decoding amedia signal comprising a stream representing encoded secondary mediadata using a reference pattern and a plurality of data words. Thedecoder 50 comprises a reference pattern analyzer 60, a signalmanipulator 65, and a signal processor 70. The reference patternanalyzer 60 is configured to analyze the reference pattern of theencoded secondary media data, wherein the reference pattern analyzer 60is configured to determine an amplitude of the reference pattern or todetermine a predetermined timing instant in the primary media data. Thesignal manipulator 65 receives the encoded secondary media data 55 andthe analyzed reference pattern 75 of the reference pattern analyzer 60.The signal manipulator 65 is configured to manipulate the encodedsecondary media data 55 in accordance with the analyzed referencepattern 75 and a computed reference pattern to obtain secondary mediadata 80. The media data, e.g. the data words, may be transmittedseparately to the signal manipulator or the media data may betransmitted directly to the signal manipulator through the referencepattern analyzer. The signal processor 70 receives the secondary mediadata 80 and is configured to process the primary media data 90 accordingto the encoded secondary media data 55 to obtain a decoded media signal85.

The media signals will be specified in the further description,especially with respect to FIGS. 8-12. According to embodiments, theencoded secondary media data is pulse code modulated (PCM) comprisingpulse amplitude modulated (PAM) symbols in the data words. To obtain thePCM modulated encoded secondary media data, the data words may be PCMmodulated.

FIG. 7 shows a schematic block diagram of the decoder 50 according to anembodiment. Herein, embodiments of the reference pattern analyzer 60 andthe signal manipulator 65 are shown.

Embodiments show the reference pattern analyzer 60 comprising a gainfactor calculator 94 configured to calculate an amplification orattenuation factor according to the amplitude of the reference patternand the amplitude of the computed reference pattern. Furthermore, thesignal manipulator 65 comprises a multiplier 27 configured to amplify orattenuate the encoded secondary media data according to theamplification or attenuation factor to obtain gain compensated encodedsecondary media data 95. Therefore, the reference pattern analyzer 60may further comprise an amplitude detector 20 configured to determinethe amplitude of the reference pattern. However, the amplitude of thereference pattern may be compared to a known amplitude of the referencepattern to obtain a gain factor. This method may work for DC-free or, inother words, with gain compensated signals. Therefore, the embodimentshown in FIG. 7 proposes a further gain calculation method bysubtracting a positive amplitude in the reference pattern and a negativeamplitude in the reference pattern using a subtractor 24 and calculatinga fraction of a known difference between the amplitudes and thecalculated difference of the amplitudes to obtain the amplification orattenuation factor.

Embodiments further show the reference pattern analyzer 60 comprising anamplitude detector 20 configured to determine the amplitude of thereference pattern and a further amplitude of the reference pattern,wherein the reference pattern analyzer further comprises an offsetcompensation unit 96 configured to calculate an offset 96 a of theencoded secondary media data 55 according to a drift of the amplitude ofthe reference pattern and a further amplitude of the reference pattern.The signal manipulator 65 therefore comprises an adder configured to addthe offset of the encoded secondary media data to the encoded secondarymedia data to obtain offset compensated encoded secondary media data 97.The drift may be calculated by adding the (positive) amplitude of thereference pattern and the (negative) further amplitude of the referencepattern. The offset, or according to embodiments, one half of theoffset, may be subtracted by subtractor 26 from the encoded secondarymedia data 55 to obtain the offset compensated encoded secondary mediadata 97.

In other words, FIG. 7 shows the conceptual operation of a receiverportion of an embodiment of the invention. The AES3 PCM audio data(secondary media data) 55 de-embedded from an SDI video signal (primarymedia data) is input to a reference amplitude detector 20, whichaverages the central four samples of the PCM audio signal during a pulseperiod 41 and during a pulse period 42 in the reference pattern 135 (cf.FIG. 7). This may be done using timing circuits based on the verticalsync signal 160 or in an alternate embodiment on a combination of thevertical sync signal 160 and an examination of the incoming PCM valuesto detect the leading edge of pulse 41 in the reference pattern 135. Themean amplitude of the pulse 41 is thus stored in a register 21 and themean amplitude of pulse 42 is similarly stored in a register 22. Theoutputs of the registers 21 and 22 are added to determine the zero levelof the original signal and input to a subtractor 26, which removes anyDC offset 96 a from the signal. The outputs of the registers 21 and 22are subtracted by a subtractor 24 to determine the peak to peakamplitude of the two pulses 41 and 42. This amplitude is fed to functionblock 25, which computes an appropriate gain factor 94 a to be appliedto a multiplier 27 to normalize the output of the subtractor 26 suchthat the original PCM signal values are nearly reproduced at the outputof the multiplier 27. Such functions as described herein will befamiliar to those skilled in the art of analog television systems designas a digital implementation of a tri-level sync detector andsync-tip-controlled AGC (automatic gain control) function.

Although the operations of the functions of 20, 21, 22, 23, 24, 25, 26,27 would ideally restore the exact values of the PCM signal (stream) 145created at the output of the transmitter filter 15 in an encoder (cf.FIG. 4), rounding errors in arithmetic operations, and ringing or otherdegradation of the pulses 41 and 42 may cause the output of themultiplier 27 to only approximate the signal produced at the filter 15.This error is reduced by averaging the four central samples of thepulses 41 and 42 in the reference pattern and by using PCM values ofsufficient size such that such approximation error does not appreciablyaffect symbol decisions as described below.

Additionally, the assignment of symbols to PCM values as shown in FIG.8c allows for amplification of the transmitted PCM signal by up to fourbits or approximately 24 dB, and also allows for a similar attenuationof four bits or approximately 24 dB, while still maintaining three LSBsas margin for rounding error or degradation of the signal.

According to further embodiments, the signal manipulator 65 comprises asample rate converter 28 configured to convert a sample rate associatedwith the digital words 140, according to a predetermined timing instantof the primary media data indicated in the reference pattern 135, to apredetermined sample rate to obtain resampled digital words. In otherwords, the received reference pattern may comprise a specific sequence,e.g. a zero crossing between two pulses, wherein the original sequencebefore transmission is known by the decoder. The decoder can calculate,based on an accurate analysis of the position of the zero crossing, adifference between the sample rate of the stream of digital words beforetransmission and after receiving the stream of digital words. Thedifference may be used to decode the stream of digital words using theoriginal sample rate of the data words before transmission.

Embodiments further show the reference pattern analyzer comprising atiming instant determiner 32 configured to determine the predefinedtiming instant of the primary media data in the reference pattern interms of samples of a sample rate, an upsampler 33 configured toupsample a range around the determined timing instant to determine anexact position of the timing instant, and a sampling accumulator 34configured to determine an exact position of the plurality of digitalwords within the stream of digital words to obtain an actual sample rate92 associated to the digital words being different from a predeterminedsample rate.

It has to be noted that according to embodiments, the predeterminedtiming instant of the primary media data is indicated as a zero-crossingbetween a positive amplitude of the reference pattern and a negativeamplitude of the reference pattern, indicating that a synchronizationsignal in the primary media data was sent before the positive amplitudeof the reference pattern. Therefore, the reference pattern analyzer isconfigured to find the zero-crossing in timing instant determiner 32.The upsampler 33 is configured to N-times upsample the area between thesample before the zero-crossing and the sample after the zero-crossing.Therefore, values of the two samples are obtained and the value of oneof the N-values between the two samples closest to zero is obtained forthe current and a following reference pattern. The sampling accumulator34 calculates the sample rate between the reference pattern and thefollowing reference pattern or, in other words, calculates that point intime corresponding to the samples in the encoded secondary media data,where the value of the current symbol may be obtained withoutinter-symbol-interference, for example due to a Nyquist filtering of theencoded secondary media data in the encoder. Therefore, the sample rateconverter 28 is configured to sample the encoded secondary media dataaccording to the calculated predetermined timing instants or the actualsample rate 92 of the sampling accumulator 34.

In other words, FIG. 7 further shows a schematic conceptual blockdiagram of a receiver portion according to embodiments that accepts PCMdata from a 16 bit, 48 kHz PCM audio channel and decodes the embedded 32PAM signal 145 into a metadata or control data bitstream according to anembodiment. To recover the PAM symbols from the normalized PCM data atthe output of the multiplier 27, the data should now be sampled atinstants corresponding to the center of the symbol period to avoidinter-symbol-interference. This is accomplished as follows: The outputof the multiplier 27 is input to a function block 32, which operates ina similar manner to the function of the detector 20 and the registers 21and 22, and outputs to a block 33 the PCM values of the normalized PCMsignal output by multiplier 27 which occur at the zero-crossing betweenpulses 41 and 42 of the reference pattern.

The function block 33 takes these two PCM values and computes the commonalgebraic formula for calculating the y-intercept of a linear functionas follows:

${f\left( {x_{n},x_{n + 1}} \right)} = {\frac{{- x_{n}} \cdot N}{x_{n + 1} - x_{n}}.}$

x_(n) is the value of the sample left from the zero crossing and x_(n+1)is the value of the sample right from the zero crossing. Thus, it can bedetermined in which of N subdivisions of a sample period thezero-crossing of the waveform represented by the PCM samples wouldoccur. In the case of this embodiment, N is set equal to 16, though thechoice of N is an engineering compromise between increased symbolsampling accuracy and the need to store additional filter coefficientsfor filter 28 as will be explained below. According to furtherembodiments, N is set equal to 128 or 256. Any other values may besuitable as well.

The combination of the accumulator 34 and the sampling/interpolatingfilter 28 is used to resample the input signal from the multiplier 27 attime instants close to the center of the symbol period. The accumulator34 functions as a fractional accumulator similar to a DDA (digitaldifferential analyzer) such as described in “Principles of InteractiveComputer Graphics”, Newman and Sproull, 2nd ed., Mc-Graw-Hill, 1979,FIG. 2-9, and is similar to phase accumulators used in digital phaselock loop design and direct digital frequency synthesizers.

In this case, the accumulator 34 is initialized with the zero-crossingsubdivision number computed by the function block 33 and thenincremented by one-half of the symbol period, which in this case is 1.5samples of the 48 kHz clock for a 16 kbaud symbol rate, to move theaccumulator location from the symbol edge to the symbol center. Theaccumulator 34 is then incremented by 1.0 for each sample clock and itsfractional bits (log 2 N) select a phase of interpolating filter 28,e.g. a polyphase FIR interpolating filter bank. The system of 34 and 28forms a sample rate converter similar to that described in “A flexiblesampling-rate conversion method,” Julius O. Smith and P. Gossett, IEEEInternational Conference on ICASSP 1984, pp. 112-115, March 1984. Thedesign of one approach of the polyphase filters is described in theabove paper.

The output of the filter 28 will then contain, at each clock cycle wherethere is a carry-out from the fractional part of the accumulator 34, amid-point sample of each received symbol. Upon such carry-out of thesampling accumulator 34, the register 29 is enabled to store the symbol,which is then input to the function block 30, which right-shifts the16-bit value seven bits with rounding, to recover the transmittedsymbol. The value of the five lower bits is then decoded from Gray codeand stored in an output buffer 31. The contents of the buffer 31 arethen available as the received audio control data or metadata bitstream(e.g. the bitstream of secondary media data 125).

The operation of the accumulator 34 as described above results inadjustment of the symbol sampling phase based solely on the timingreference from the pulses 41 and 42 sent after each vertical sync pulse.It will be understood by those skilled in the art that this will correctphase errors between the incoming symbols and the local symbol samplingclock, but might not completely correct any frequency error. With thedesign goals above, a 100 ppm frequency error in the transmittertime-base will result in a sample error of 0.15 of a sample clock or0.050 of the symbol width at the very end of a data payload just beforethe vertical sync interval.

This error could be further reduced by adding a frequency term to theincrement of the accumulator 34. Such a term may be calculated bycomparing the fractional part of the accumulator with the value to whichit is to be initialized following the vertical sync period. Thisdifference of these values can then be divided by the approximate orexact number of sample clocks since the last vertical sync period andadded to the 1.0 value used to increment the sampling accumulator 34. Inthis manner, most of the effect of a frequency error may be removed.

According to a further embodiment, the signal manipulator comprises ademapper 29 configured to demap grouped secondary media data comprisinga first number of bits from the data words comprising a second number ofbits being greater than the first number of bits. Additionally oralternatively, the signal manipulator comprises an ungrouper 31configured to ungroup grouped secondary media data comprising a firstnumber of bits to obtain a decoded bitstream of secondary media data125′, which is a bitstream representation of the secondary media data 80and therefore represents the bitstream of secondary media data 125.

The following FIGS. 8 to 12 describe embodiments of encoded secondarymedia data, indicating that the data words are PAM coded and that theapplication to the (Nyquist) filter 15 results in a PCM signal.

FIG. 8a shows a schematic representation of the media signal 155according to an embodiment. The media signal comprises a stream ofdigital words 145 representing encoded secondary media data 55comprising metadata or control data for primary media data.

FIG. 8b shows a schematic representation of the media signal 155according to a further embodiment. The media signal comprises a stream145 representing encoded secondary media data 55 using a referencepattern 135 and a plurality of data words 140, wherein the plurality ofdata words comprise secondary media data. Furthermore, the encodedsecondary media data is mapped into the plurality of data words with agap to the most significant bit of the data word or the leastsignificant bit of the data word. According to embodiments, thereference pattern 135 and the data words 140 are filtered to derive thedigital words 142, or more precisely, the stream of digital words 145.

The reference pattern comprises the same structure as the data words140, meaning that the bitstream of secondary media data 125 comprises areference pattern 135, which is grouped into a grouped reference pattern(according to the grouped secondary media data) and formed in a dataword such as the data word 140. This would result in a uniformprocessing within the encoder 100 shown e.g. in FIG. 4, wherein switch13 is configured to switch between the reference pattern 135 and themetadata or control data of the primary media data. In other words, thesecondary media data comprises the grouped reference pattern andmetadata or control data for the primary media data in a firstembodiment. In a second embodiment, the reference pattern is independentfrom the secondary media data. The differentiation is advantageous sincethe processing of the reference pattern and the metadata or control datais optionally joint or separate from each other. Furthermore, thedecoded media signal 85 or the decoded bitstream of secondary media data125′ is ideally identical or at least similar in terms of e.g. roundingerrors to the encoded bitstream of secondary media data 55.

Embodiments show the reference pattern 135 comprising a referenceamplitude of the encoded secondary media data and a predetermined timinginstant in primary media data. According to further embodiments, themedia signal comprises a further stream of the primary media data,wherein the primary media data comprises audio data or video data. Thefurther stream comprising primary media data is aligned to the stream ofencoded secondary media data at the predetermined timing instant in theprimary media data. The primary media 90 a or 90 b comprises the timinginstant 40 being represented in the reference pattern e.g. by the zerocrossing 165.

FIG. 8c shows a schematic representation of the data word 140 accordingto an embodiment. The grouper groups the bitstream of secondary mediadata into grouped secondary media data 130 comprising five bits (e.g.bits 7 to bit 11), wherein the mapper is configured to sign extend 130 athe grouped secondary media data to the most significant bit (forexample bits 12 to 15), meaning that the first bit (bit 11) of thegrouped secondary media data is padded to the bits 15 to 12, and whereinthe mapper further pads the gap to the least significant bits (e.g. bits6 to 0) with zeros 130 b. Further embodiments show the secondary mediadata comprising eight bits. The padding to the left or to the right isreduced accordingly by 3 bits in total to obtain a 16 bit data word.Other combinations such as a different length of the secondary mediadata or the data word or another size of the padding may be alsorealized. Furthermore, the reference pattern may be processed such thatthe reference pattern comprises the same structure as the data word 140.

FIG. 9a shows a timing instant 40 in the primary media data 160indicating, for example, a vertical blanking interval, or a furthersynchronization point in the video frame. Advantageously, thesynchronization part 40 indicates a suitable point of time in a videoframe which indicates a suitable position to cut a stream of videoframes. This might be the vertical blanking interval or for example acertain line in the video frame (e.g. line 7), where cutting of a videostream may be performed. Therefore, the distance between two consecutivesynchronization pulses is one frame. One frame may comprise 800 or 801audio samples, which results in around 300 data words per video frameand additional reference pattern, version number, continuity counter,cyclic redundancy check or further overhead.

FIG. 9b shows a schematic representation of the stream 145 representingencoded secondary media data using a reference pattern and a pluralityof data words. Since FIG. 9b is aligned to FIG. 9a , it is shown thatthe reference pattern 135 is driven by the timing instant 40. Therefore,the predetermined timing instant 165, being the zero crossing betweenamplitudes 41 and 42 of the reference pattern according to thisembodiment, indicates the timing instant 40 in the synchronizationsignal 160 of the primary media data. The first amplitude of thereference pattern 41 may comprise an amplitude of 0x0780 HEX, whereinthe second amplitude 42 of the reference pattern may comprise a value of0xf880 HEX. Adjacent to the first and second amplitude of the referencepattern, it may be padded with zeros or, according to furtherembodiments, the zero padding is part of the reference pattern. Afterthe reference pattern is processed, the stream builder applies the datawords 140 to the data payload container 43. Further embodiments show anadditional part in the payload container 43, where redundancy is appliede.g. to perform bit error corrections like checksums, parity bits,cyclic redundancy checks, etc. The reference pattern 135 and the datawords 140 may be filtered to obtain digital words 142 to form the stream145.

The following FIGS. 10 to 12 describe the filter 15, the stream builder120, and the stream 145 in more detail. FIG. 10a shows a raised cosineshape filter with a rolloff factor=0.98, wherein FIG. 10b shows theraised cosine shape filter sampled according to a sampling frequency. Itmay be seen that the raised cosine shape filter having a rolloff factorof 0.98 puts almost all of the energy of the impulse in the three middlesamples 180 a, 180 b. However, there may be used 13 samples for theaddition or more precisely only the seven coefficients that aredifferent from zero. Using only the three middle samples, however, willalso enable a good reconstruction of the encoded symbol without aliasingproblems or inter-symbol-interference.

FIGS. 11a and 11b show the raised cosine shape filter function 15′ witha rolloff factor 0.7 in a time-continuous representation (FIG. 11a ) anda time-discrete representation (FIG. 11b ). FIG. 11c shows the image ofFIG. 11b three times in a row, aligned with an offset of two samplesbetween consecutive filter functions, which may be the data pulse 15′.The filter functions or the data pulses 15′ are modulated, e.g.multiplied, with the mapped secondary media data (representing onesymbol of secondary media data) or (a symbol) of the reference pattern,each representing a data word 140 or a (PCM modulated) symbol of areference pattern. The parameters are chosen in such a way that everysecond sample of the discrete representation of the raised cosine filteris zero. Therefore, two adjacent pulses are placed with a distance oftwo samples, such that the middle of each pulse is at a position whereall other pulses are crossing zero. This concept is quite simple for themodulation process and also simple for the demodulation, where examiningthe middle sample comprises the compensation for timing errors and gainerrors. If a clock deviation, or a difference between an originalsampling frequency and an actual sampling frequency, of the digitalwords after transmission is sufficiently low, a symbol recovery in thedecoder may be performed without calculating the source samplingfrequency. Furthermore, a small number of amplitude values is beneficialfor symbol recovery without sample rate conversion in the decoder.However, it may be advantageous to apply a phase compensationindependently from a correction of the clock deviation.

An addition of the values of each sample (from top to bottom) results inthe stream 145 of digital words. Furthermore, the amplitude or, in otherwords, the values of each sample are weighted (e.g. multiplied) with thedata word 140 or the symbol of the reference pattern, which may be seenas a pulse amplitude modulation. These schematics are applied to thereference pattern and the data words according to embodiments.Furthermore, it has to be noted that the embodiments described with24000 symbols per second and 256 amplitude values (8 bit) or 32amplitude values (5 bit) are exemplary and not limiting the scope of theinvention. Other symbol rates are conceivable, both lower and highersymbol rates using sample rate conversion to insert the symbols at zerocrossings of the stream comprising secondary media data as well asdifferent resolutions for the amplitude steps.

FIG. 12 shows a schematic representation of the stream 145 according toan embodiment. FIG. 12a shows a schematic time-continuous representationof the stream 145 comprising the filtered reference pattern 135 and thefiltered data word 140. Furthermore, a second reference pattern 135 a isshown, which may be optionally applied at the end of the frame toachieve an accurate timing recovery within a signal frame. Therefore,the second synchronization symbol (or reference pattern) 135 a mighthave a slightly lower amplitude than the first synchronization symbol135 and furthermore, the first synchronization symbol 135 might comprisea higher amplitude than all of the other symbols. In that way, it isvery efficient to search for the first synchronization symbol.Furthermore, the data word may comprise one or more redundancy bits toenable an error detection. FIG. 12b shows the stream 145 in an enlargedversion. FIG. 12c shows a signal similar to the signal shown in FIG. 12ain a time-discrete form at samples of a sample rate. Furthermore, FIG.12d shows a signal similar to the signal shown in FIG. 12b in atime-discrete form.

FIG. 13 shows a schematic flow diagram of a method 1100 for decoding amedia signal comprising a stream representing secondary media data usinga reference pattern and a plurality of data words, the method 1100comprises a step 1105 for recovering the secondary media data with adecoder, the recovering comprising manipulating the received stream ofdigital words with respect to amplitudes represented by the receiveddigital words or using resampling, and step 1110 for deriving abitstream from the recovered secondary media data.

FIG. 14 shows a schematic flow diagram of a method 1200 for encoding amedia signal with an encoder. The method 1200 comprises a step 1205 forencoding the secondary media data with an encoder using addingredundancy or bandlimiting and a step 1210 for outputting the encodedsecondary media data as a stream of digital words.

Construction Considerations of an Embodiment

The described embodiments may be implemented in software as a series ofcomputer instructions or in hardware components. The operationsdescribed here are typically carried out as software instructions by acomputer CPU or Digital Signal Processor and the registers and operatorsshown in the figures may be implemented by corresponding computerinstructions. However, this does not preclude embodiments in anequivalent hardware design using hardware components. Further, theoperation of the invention is shown here in a sequential, elementarymanner. It will be understood by those skilled in the art that theoperations may be combined, transformed, or pre-computed in order tooptimize the efficiency when implemented on a particular hardware orsoftware platform.

Alternate Embodiment for Audio-Only Systems

The invention may be furthermore used in audio-only system withoutdistributed vertical sync by replacing the vertical sync signal in thetransmitter by an equivalent locally generated signal, and by protectingthe data bitstream input to register 11 from symbol patterns that willgenerate pulses identical to pulse 41, through convolutional coding orother means. Reference Amplitude Detector 20 may then be modified toregenerate a local sync signal in the receiver by detection of pulse 41.

In a further embodiment, the modulation for the audio metadata which areprovided as a stream of bits to obtain an audio-like digital stream,such as a stream at the output of block 3 in FIG. 1 may comprise severalprocedures alternatively to each other or in addition to each other. Inparticular, the stream output by block 3 in FIG. 6 and input into block4 in FIG. 6 is, for example, a sequence of PCM values such as 16 bits or32 bits PCM values such as those which are, for example, stored on a CD.Naturally, the control data or metadata bitstream has a certainbitstream syntax and the actual digital words consisting of several bitsin the metadata bitstream will typically have variable lengths. However,the block 3, or generally a procedure for generating an audio-likedigital stream from the audio control data or metadata comprises agrouper for grouping a first number of bits from the stream. Thus, thismeans, for example, that a sequence of 5 bits is taken from the metadatabitstream. Then, a state represented by the first number of bits, i.e.by 5 bits, is determined. This state is one of 32 states. Then, in oneembodiment, the state is represented by a second number of bits, wherethe second number of bits is greater than the first number of bits. Thisrepresentation into the second number of bits can, for example, be a 16bits representation or a 32 bits representation or so. In any case,however, the second number of bits is greater than the first number ofbits so that a certain kind of robustness or redundancy is introducedinto the representation. Then, the state represented by the secondnumber bits is written into a sequence of digital words all consistingof the second number of bits and this writing is performed a single timeor, in order to even increase the redundancy, more than one time in thesequence. Advantageously, the state is written into the sequence two,three or even more times in sequence so that the audio-like digitalstream generated by this embodiment is a stair-like form having a groupof identical values followed by another group of identical values andthe height or state of these values is only one of a certain number ofstates, such as only one of the 32 different possible states, althoughthe individual values are not represented by, for example, 5 bitsvalues, but are represented by 16 or 32 bits values. Alternatively, acertain redundancy is already obtained by grouping into the first numberof bits and by then writing the first number of bits into the sequenceof digital words more than one time in sequence, i.e. by a repetition ofa certain number of times.

Depending on the applied redundancy, i.e. a redundancy by having asecond number of bits being greater than a first number of bits and/orby repeating the state a certain number of times, different ways ofreconstruction on the receiver-side can be performed. For example, whenonly a kind of repetition is performed, then the for example threesubsequent values which should be same are taken and a decision isperformed saying that the value is the value which is represented by twoor those three values. Thus, a majority decision can be taken.

Alternatively or additionally, and particularly when the embodiment withthe second number of bits being greater than the first number of bitshas been applied, i.e. when a 5 bit state, for example, is representedby 16 bits, in addition to a majority decision or as a furtheringredient of the decision or instead of the majority decision, alow-pass filtering or a mean value calculation or a so can be performedin order to find out or reconstruct the original value.

The inventive transmitted or encoded signal can be stored on a digitalstorage medium or can be transmitted on a transmission medium such as awireless transmission medium or a wired transmission medium such as theInternet.

Embodiments show a different PCM channel for the metadata or controldata, allowing the essence audio signals (or primary media data) to betransmitted with full quality and resolution. Furthermore, the controldata or metadata signal may be transformed into one that can survivetypical degradations of PCM audio signals, such as gain changes, timebase errors, resampling, changes in delay relative to the primarysignal, etc. Moreover, embodiments may operate in the advantageous, butnot exclusive, case with unencoded or uncompressed essence signals.

Further embodiments are described below:

The new MPEG-H based TV audio system will bring three primary newfeatures to television broadcasts. “MPEG-H” refers to part 3 of theMPEG-H standard, ISO/IEC 23008-3, and may not relate to the other partsconcerned with MMT transport, HEVC video coding, etc. More specifically,to the new TV Audio System developed by the MPEG-H Audio Alliance basedon the MPEG-H Audio codec. The three primary new features are:

-   -   Interactivity to enable consumers to choose different audio        presentations, such as a home team or away team commentary at a        sports event, or to turn up or down particular audio elements in        a program—such as dialogue or sound effects—as they like.    -   Immersive sound to improve the realism of the sound by adding        height channels, using MPEG-H's Higher-Order Ambisonics mode, or        statically panned objects above the listener.    -   Multi-platform Adaption. Unlike today's TV sound, the MPEG-H        system will tailor playback so it sounds best on a range of        devices and environments—from quiet home theaters with speakers        to the subway or airport with earbuds.

All of these features will be under the control of the broadcaster orcontent distributor, providing new creative opportunities, such as theability to efficiently add additional languages, player, or officialmicrophones, or, as the Alliance has demonstrated, car to pit crewradios at races.

Since the MPEG-H Audio system is designed to work over unmodified HD-SDIembedded audio channels, stations can begin implementing MPEG-H Audiofeatures as they choose without changing their internal plant oroperating procedures. A four-stage process for broadcasters to considerwhen adopting MPEG-H is proposed:

-   -   1. Transmission of stereo and surround programming using MPEG-H        Audio: This would allow broadcasters to gain the bitrate        efficiency and new mobile audio features of MPEG-H Audio without        any operational changes.    -   2. Addition of audio objects for additional languages or        alternate commentary, enabling viewers to Hear Your Home Team™        audio or listen to their favorite race driver's radio, as well        as providing for mandated access features such as visual        description.    -   3. Addition of immersive sound to improve the realism of the        sound by adding height channels, Higher-Order Ambisonics, or        statically panned objects above the listener.    -   4. Addition of dynamic audio objects: In contrast to static        objects fixed in position, dynamic objects move over time to        track video action or provide creative effects. If sound effects        are to be panned, for example, a dynamic object can reduce the        bitrate that may be useful compared to sending a five or nine        channel static object.

Adapting live production and playout for MPEG-H: two approaches

In today's television plants, live or real-time video signals aretransported using the HD-SDI interface which supports up to 16 channelsof embedded audio. An exemplary system is designed to use these channelsdirectly for the channels, objects, and other audio elements of aprogram.

FIG. 15 shows a schematic diagram of a MPEG-H distribution systemaccording to an embodiment, where FIG. 15a shows the system in a fixedmode and FIG. 15b shows the system in a Control Track Mode. For stages 1to 3 above, the traditional approach (c.f. FIG. 15a ) of using a fixedchannel map or rundown and fixed encoding metadata may be used. Thisapproach has the advantage of being easy to understand, and may use verylittle in terms of operational changes if objects are not used or only afew routine objects are used. This approach is termed the Fixed Mode,although presets can be used under external control to change theencoder settings.

The fixed mode represented by FIG. 15a basically shows an MPEG-H AudioMonitoring and Authoring Unit 200 which may be operated in monitoringmode. Input to the Monitoring and Authoring Unit 200 is the video withembedded audio 205 such as the HD-SDI signal comprising up to 16 audiochannels. The MPEG-H Audio Monitoring and Authoring Unit 200 may beconfigured to use a web-based control interface 210, which sets fixedpresets for channel assignment and audio parameters. Output of theMPEG-H Audio Monitoring and Authoring Unit 200 is a remote control 215comprising monitor controls 220 and integrated loudness instruments 225.The web-based control interface or the remote control (or both) may beconnected to the MPEG-H Audio Monitoring and Authoring Unit 200 by aninternet protocol connection 240. Furthermore, the MPEG-H AudioMonitoring and Authoring Unit 200 may be connected to speakers (notshown) using connection 235.

The HD-SDI signal 205 is input to a Video/MPEG-H Audio Contribution orDistribution Encoder 245 comprising a video encoder 250 and an MPEG-Hencoder 255. The MPEG-H encoder may be fed with fixed presets forchannel assignment and audio parameters using the web-based controlinterface 210 and the internet protocol connection 240. The output ofthe video encoder 250 and the MPEG-H encoder 255 is input to a transportmultiplexer 260. The multiplexed signal 265 is distributed ortransmitted using e.g. internet protocol (IP) or digital videobroadcasting asynchronous serial interface (DVB/ASI).

A Video/MPEG-H Audio Contribution or Distribution Decoder 270 receivesthe multiplexed signal 265 and a transport demultiplexer 275demultiplexes the multiplexed signal 265. The demultiplexed signal maybe fed into a video decoder 280 and a MPEG-H decoder 285 forming adecoded version 205′ of the video signal with embedded audio comprisingup to 16 channels 205. Further audio processing applied to the decodedsignal 205′ may be equivalent to the processing of the audio signals inthe HD-SDI video signal 205 before transmission.

According to an embodiment, an alternative approach, the Control TrackMode (cf. FIG. 15b ), was developed, which uses a Control Track placedon one of the audio channels, usually channel 16. The control track maycomprise the metadata or control data for primary media data.

The schematic block diagram presented in FIG. 15b shows a few changescompared to the block diagram described with respect to FIG. 15a . Firstof all, the MPEG-H Audio Monitoring Unit 200 operates in authoring mode,which enables the monitoring unit 200 to generate the control track andinsert the control track e.g. into channel 16 of the video with embeddedaudio comprising up to 15 channels. The 16^(th) channel might remain forthe control track. Channel assignment and audio parameters forgenerating the control track may be set by a web-based control interface210. The further processing of the video signal with embedded audiocomprising up to 15 audio channels and the generated control track 205″is similar to the signal processing in FIG. 15a . However, channelassignment and audio parameters are read from the control track and donot need to be applied using e.g. a web interface.

The Control Track may be synchronized to vertical sync to allow easyvideo editing and switching. The Control Track is designed to operatejust like a longitudinal time code signal. It will survive normalprocessing of a PCM audio channel, but it cannot be successfullytransmitted over a compressed audio channel such as a Layer IIcontribution codec. For this situation, an MPEG-H Audio contributionencoder may be used, which compresses the audio channels fortransmission and converts the control track into metadata carried in theMPEG-H Audio bitstream.

The Control Track:

-   -   contains all the configuration information needed by the        encoder, including        -   channel map or rundown        -   object names or labels        -   object groups and control limits        -   program reference level (“dialnorm” in the MPEG            terminology), downmix gains, and DRC profiles        -   position information for dynamic objects    -   may be switched in routing, production, or master control        switchers    -   will pass through frame synchronizers and other terminal        equipment    -   may be edited with the other audio tracks in a video editor or        audio workstation    -   will pass through an audio console with the other audio tracks    -   provides frame-accurate transitions of the encoded or monitored        audio to match video program switches or edits    -   does not involve configuring equipment for “data mode” or        “non-audio mode” treatment of the control track channel

The Control Track, since it is carried in an audio channel with thecontent, provides automatic setting of all parameters of the MPEG-HAudio Encoder without any manual programming or need to modify otherequipment in the plant. The Encoder translates the Control Trackinformation into MPEG-H audio metadata which is transmitted in theencoded bitstream to the MPEG-H Audio Decoder. This mode of operation istermed the Control Track Mode.

Professional Decoders may be operated in a contribution or transmissionmode, where they recreate the Control Track signal from the receivedmetadata, or in an emission mode where they render the audio channelsjust as a consumer decoder would.

The Control Track may be generated by the Audio Monitoring and AuthoringUnit used by the audio operator for a live program. For ingest ofrecorded content, either the HD-SDI signal may be passed through anAudio Monitoring and Authoring Unit for adding the control track duringreal-time dubbing, or file-based utilities may be used to insert thecontrol track into common file formats such as QuickTime/MP4FF or MXF.Of course, the Audio Monitoring and Authoring Unit also uses the ControlTrack during monitoring to simulate the actions of the MPEG-H AudioDecoder.

Since the control track may be edited just like any other audio channel,programming with different channel assignments or different objects canbe combined in an editor just by dropping items on the editing timeline.

Use of the Control Track means one audio channel is no longer availablefor objects or channels, but also opens the possibility of using dynamicobjects. For panned sounds, such as sound effects, several channels ofstatic objects could be useful to create the effect that may be donewith a single-channel dynamic object.

The Control Track approach allows full flexibility in the MPEG-H audiomodes used during a broadcast day. It is easily possible to have a showwith a stereo bed and two dialogue objects be interrupted by programinserts in full immersive 7.1+4H sound, or even Higher-Order Ambisonics,interspersed with commercial breaks in stereo or 5.1 surround.

One new possibility shown is the ability to broaden the reach ofcommercials to include demographics who are more comfortable listeningto advertisements in their primary language. Local spots intended toreach the broadest possible audience could have voiceovers or dialog inseveral languages selected by the advertiser. The Preferred Languagefeature of the exemplary system will present the commercial the viewerspreferred language if broadcast, and automatically switch back to thedefault language for other programming or commercials that do not havethat language present.

With certain restrictions on content transitions, primarily duringnetwork break and join operations, it is possible to have a mixture ofnew content with the Control Track signal and legacy content without.For example, the MPEG-H Audio Encoder and MPEG-H Audio Monitoring andAuthoring Unit can be set to switch to 5.1 surround mode with a fixedloudness of −24 LKFS (Loudness, K-weighted, relative to Full Scale) andstandard downmix gains and DRC profiles, as a facility typically usestoday. In this manner, legacy content would be encoded as it is today,and new content with immersive or interactive features wouldautomatically be encoded with the correct settings.

Further embodiments of the invention relate to the following examples:

1. A system for transmitting or receiving data in a digital audiochannel by digitally modulating or encoding said data into a signalbandlimited or tolerant of transmission degradations for transmission insaid channel, or a signal that is not raw bits somehow packed together,but survives channel degradations.

2. The system of example 1 where the data is control data, metadata, orother data relating to an audio signal carried in a second digital audiochannel.

3. A system for transmitting a data-compressed digital audio bitstreamin a digital audio channel by digitally modulating or encoding saidbitstream for transmission in said channel.

4. The system of example 3 where the data-compressed digital audiobitstream contains metadata or control data

5. The system of example 3 where the data-compressed digital audiobitstream only contains metadata or control data and not the relatedaudio information.

6. The system of example 1 where said digital audio channel is embeddedinto a digital video signal.

7. The system of example 2 where said digital audio channel is embeddedinto a digital video signal.

8. The system of example 3 where said digital audio channel is embeddedinto a digital video signal.

9. The system of example 4 where said digital audio channel is embeddedinto a digital video signal.

10. The system of example 5 where said digital audio channel is embeddedinto a digital video signal.

11. Method, apparatus or computer program for modulating audio controldata or metadata comprising a stream of bits to obtain an audio-likedigital stream, comprising:

grouping a first number of bits;determining a state represented by the first number of bits;representing the state by a second number of bits, the second number ofbits being greater than the first number of bits and writing the secondnumber of bits into a sequence of digital words consisting of the secondnumber of bits a single time or more than one time in sequence; orwriting the first number of bits into a sequence of digital words morethan one time in sequence.

12. Method, apparatus or computer program for demodulating a digitalstream to obtain a stream of bits of audio metadata or control data,comprising:

performing a majority decision or a mean value calculation between asequence of received audio samples to obtain a grouped first number ofbits or a quantization of an audio sample into a number of bits; andsyntactically parsing a sequence of bits obtained by concatenating twoor more groups of a first number of bits to obtain the metadatainformation.

Although the present invention has been described in the context ofblock diagrams where the blocks represent actual or logical hardwarecomponents, the present invention can also be implemented by acomputer-implemented method. In the latter case, the blocks representcorresponding method steps where these steps stand for thefunctionalities performed by corresponding logical or physical hardwareblocks.

Although some aspects have been described in the context of anapparatus, it is clear that these aspects also represent a descriptionof the corresponding method, where a block or device corresponds to amethod step or a feature of a method step. Analogously, aspectsdescribed in the context of a method step also represent a descriptionof a corresponding block or item or feature of a correspondingapparatus. Some or all of the method steps may be executed by (or using)a hardware apparatus, like for example, a microprocessor, a programmablecomputer or an electronic circuit. In some embodiments, some one or moreof the most important method steps may be executed by such an apparatus.

Depending on certain implementation requirements, embodiments of theinvention can be implemented in hardware or in software. Theimplementation can be performed using a digital storage medium, forexample a floppy disc, a DVD, a Blu-Ray, a CD, a ROM, a PROM, and EPROM,an EEPROM or a FLASH memory, having electronically readable controlsignals stored thereon, which cooperate (or are capable of cooperating)with a programmable computer system such that the respective method isperformed. Therefore, the digital storage medium may be computerreadable.

Some embodiments according to the invention comprise a data carrierhaving electronically readable control signals, which are capable ofcooperating with a programmable computer system, such that one of themethods described herein is performed.

Generally, embodiments of the present invention can be implemented as acomputer program product with a program code, the program code beingoperative for performing one of the methods when the computer programproduct runs on a computer. The program code may, for example, be storedon a machine readable carrier.

Other embodiments comprise the computer program for performing one ofthe methods described herein, stored on a machine readable carrier.

In other words, an embodiment of the inventive method is, therefore, acomputer program having a program code for performing one of the methodsdescribed herein, when the computer program runs on a computer.

A further embodiment of the inventive method is, therefore, a datacarrier (or a non-transitory storage medium such as a digital storagemedium, or a computer-readable medium) comprising, recorded thereon, thecomputer program for performing one of the methods described herein. Thedata carrier, the digital storage medium or the recorded medium aretypically tangible and/or non-transitory.

A further embodiment of the invention method is, therefore, a datastream or a sequence of signals representing the computer program forperforming one of the methods described herein. The data stream or thesequence of signals may, for example, be configured to be transferredvia a data communication connection, for example, via the internet.

A further embodiment comprises a processing means, for example, acomputer or a programmable logic device, configured to, or adapted to,perform one of the methods described herein.

A further embodiment comprises a computer having installed thereon thecomputer program for performing one of the methods described herein.

A further embodiment according to the invention comprises an apparatusor a system configured to transfer (for example, electronically oroptically) a computer program for performing one of the methodsdescribed herein to a receiver. The receiver may, for example, be acomputer, a mobile device, a memory device or the like. The apparatus orsystem may, for example, comprise a file server for transferring thecomputer program to the receiver.

In some embodiments, a programmable logic device (for example, a fieldprogrammable gate array) may be used to perform some or all of thefunctionalities of the methods described herein. In some embodiments, afield programmable gate array may cooperate with a microprocessor inorder to perform one of the methods described herein. Generally, themethods are performed by any hardware apparatus.

While this invention has been described in terms of several embodiments,there are alterations, permutations, and equivalents which fall withinthe scope of this invention. It should also be noted that there are manyalternative ways of implementing the methods and compositions of thepresent invention. It is therefore intended that the following appendedclaims be interpreted as including all such alterations, permutationsand equivalents as fall within the true spirit and scope of the presentinvention.

REFERENCES

-   International Organization for Standardization and International    Electrotechnical Commission, ISO/IEC 14496-3 Information    technology—Coding of audio-visual objects—Part 3: Audio,    www.iso.org.-   International Organization for Standardization and International    Electrotechnical Commission, ISO/IEC 23003-1:2007 Information    technology—MPEG audio technologies—Part 1: MPEG Surround,    www.iso.org.-   International Organization for Standardization and International    Electrotechnical Commission, ISO/IEC DIS 23008-3 Information    technology—High efficiency coding and media delivery in    heterogeneous environments—Part 3: 3D audio, www.iso.org.-   European Telecommunications Standards Institute, ETSI TS 101 154:    Digital Video Broadcasting (DVB); Specification for the use of Video    and Audio Coding in Broadcasting Applications based on the MPEG-2    transport stream, www.etsi.org.-   Advanced Television Systems Committee, Inc., Audio Compression    Standard A/52, www.atsc.org.-   Dolby Laboratories, Inc., “Post It with Dolby E”, www.dolby.com,    2003.-   Snell Ltd., “White Paper: Dolby E Processing, Working with Dolby E    in a Broadcast Environment”, www.snellgroup.com, 2011.-   A. W. J. Oomen, M. E. Groenewegen, R. G. van der Waal, and R. N. J.    Veldhuis, “A Variable-Bit-Rate Buried-Data Channel for Compact    Disc,” J. Audio Eng. Soc., vol. 43, p. 23-28 (1995    January/February).-   Audio Engineering Society, AES 3-2003, AES standard for digital    audio—Digital input-output interfacing—Serial transmission format    for two-channel linearly represented digital audio data,    www.aes.org, 2003.-   Audio Engineering Society, AES10-2008 (r2014): AES Recommended    Practice for Digital Audio Engineering—Serial Multichannel Audio    Digital Interface (MADI), www.aes.org, 2014.-   Audio Engineering Society, AES67-2013: AES standard for audio    applications of networks—High-performance streaming audio-over-IP    interoperability), www.aes.org., 2013.-   Society of Motion Picture and Television Engineers, Ancillary Data    Packet and Space Formatting, ST 291-1:2011.-   Society of Motion Picture and Television Engineers, 1.5 Gb/s    Signal/Data Serial Interface ST 292-1:2012.-   Society of Motion Picture and Television Engineers, Format for    Non-PCM Audio and Data in an AES3 Serial Digital Audio Interface ST    337:2008.-   Society of Motion Picture and Television Engineers, Format of Audio    Metadata and Description of the Asynchronous Serial Bitstream    Transport ST 2020-1:2014.-   “A flexible sampling-rate conversion method,” Julius O. Smith and P.    Gossett, IEEE International Conference on ICASSP 1984, pp. 112-115,    March 1984.-   “Principles of Interactive Computer Graphics”, Newman and Sproull,    2nd ed., Mc-Graw-Hill, 1979, FIG. 2-9,

1-45. (canceled)
 46. Encoder for encoding secondary media datacomprising metadata or control data for primary media data, wherein theencoder is configured to encode the secondary media data to obtain astream of digital words, the encoding comprising transforming thesecondary media data by a digital modulation or comprising bandlimiting,and wherein the encoder is configured to output the encoded secondarymedia data as the stream of digital words.
 47. Encoder according toclaim 46, wherein the encoding comprises adding redundancy by thedigital modulation.
 48. Encoder according to claim 46, wherein thedigital modulation is so that two or more bits of the secondary mediadata are transmitted per digital word of the stream of digital words.49. Encoder according to claim 46, wherein the encoder is configured tooutput the stream of digital words so that the stream of digital wordsis transmittable over a PCM audio channel.
 50. Encoder according toclaim 46, wherein the encoder is configured to output a further streamof digital words, the further stream of digital words representing theprimary media data, the further stream being separate from the stream ofdigital words.
 51. Encoder according to claim 50, wherein the primarymedia data are audio data, and wherein the secondary media data aremetadata for the audio data or control data for the audio data. 52.Encoder according to claim 50, wherein the encoder is configured tooutput the stream of digital words and the further stream of digitalwords so that the further stream of digital words is transmittable overa first audio PCM channel and so that the stream of digital words istransmittable over a second audio PCM channel being different from thefirst audio PCM channel.
 53. Encoder of claim 50, wherein each of thedigital words of the further stream representing the primary media datahas a predefined number of bits being greater than 8 bits and smallerthan 32 bits, and wherein each of the digital words of the stream ofdigital words has the predetermined number of bits as well.
 54. Encoderaccording to claim 46, wherein the digital modulation is a pulseamplitude modulation.
 55. Encoder according to claim 46, wherein theencoder is configured to generate the stream of digital words so thatthe stream of digital words comprises a timing reference pattern or anamplitude reference pattern.
 56. Encoder according to claim 46, whereinthe encoder is configured to output a video stream representing asequence of video images, and wherein the encoder is configured tooutput the stream of digital words so that the control data or meta dataof the secondary me dia data related to a certain video image arerelated to the certain video image.
 57. Encoder according to claim 56,wherein the encoder is configured to output the stream of digital wordsas a first stream of digital words associated to a first video image ofthe sequence of video images, and to output the stream of digital wordsas a second stream of digital words associated to a second video imageof the sequence of video images, wherein the first and second digitalwords are identical to each other.
 58. Encoder according to claim 46,wherein the encoder is configured to generate the digital words, thedigital words having 12 to 28 bits, or wherein the digital words aresampled at a sampling rate of between 30 kHz to 55 kHz, or wherein thedigital words have a dynamic range of 70 to 160 dB, or have a nominalsignal level of −20 dB RMS full scale.
 59. Encoder according to claim46, wherein the encoder is configured to use an upper frequency forbandlimiting the secondary media data being between 15 kHz to 27.5 kHzfor a sampling rate between 30 kHz to 55 kHz.
 60. Encoder according toclaim 46, the encoder comprising: a mapper configured for mapping agroup of secondary media data comprising a first number of bits into adata word comprising a second number of bits being greater than thefirst number of bits and wherein the grouped secondary media data isaligned with a gap to a most significant bit or a least significant bitof the data word; a stream builder configured for building a stream ofdigital words representing encoded secondary media data.
 61. Encoderaccording to claim 46, wherein the encoder comprises a grouperconfigured for grouping a bitstream of secondary media data to formgrouped secondary media data.
 62. Encoder according to claim 46, whereinthe encoder comprises a reference signal generator configured forgenerating a reference pattern indicating a reference amplitude or apredetermined timing instant in the primary media data; and wherein astream builder is configured to build a stream of digital wordsrepresenting encoded secondary media data using the reference pattern orthe data word.
 63. Encoder according to claim 46, wherein a streambuilder comprises a filter to low-pass filter data words or a referencepattern to obtain digital words comprising a length of more than onesample of a predetermined sample rate, wherein an amplitude of thedigital word is weighted according to the data word or the referencepattern, and wherein the filter is configured to add up consecutivedigital words at instants of the predetermined sample rate to obtain thestream of digital words.
 64. Encoder according to claim 46, wherein afilter is configured to obtain zero points at instants of apredetermined sample rate of a data pulse, wherein the data pulsecomprises a data word comprising grouped secondary media data or thereference pattern; wherein a stream builder is configured to build thestream representing the encoded secondary media data using the referencepattern and a plurality of data words such that zero points of the datapulse are aligned with a maximum of a further data pulse to obtain aninter-symbol-interference (ISI)-free stream representing the encodedsecondary media data.
 65. Encoder according to claim 46, wherein thefilter comprises a cutoff frequency of less than 1.5 times of a samplingfrequency of the primary media data.
 66. Encoder according to claim 46,wherein a reference signal generator is configured to generate a groupedreference pattern comprising a first number of bits and wherein thereference signal generator is further configured to map the groupedreference pattern into a data word comprising a second number of bitsbeing greater than the first number of bits; or wherein the mapper isconfigured to map a grouped reference pattern comprising a first numberof bits into a data word comprising a second number of bits beinggreater than the first number of bits.
 67. Encoder according to claim46, wherein the encoder is configured to output the encoded secondarymedia data as the stream of digital words as a control track and tooutput up to 15 channels of the primary media data as audio tracks,wherein the control track and the audio tracks are formed in accordancewith the AES 3 standard.
 68. Decoder for decoding a media signalcomprising a received stream of digital words representing encodedsecondary media data comprising metadata or control data for primarymedia data; wherein the decoder is configured to recover the secondarymedia data, wherein the recovering comprises applying a digitaldemodulation operation or a resampling operation to obtain recoveredsecondary media data, and wherein the decoder is configured to derive abitstream from the recovered secondary media data.
 69. Decoder accordingto claim 68, wherein the recovering comprises manipulating the receivedstream of digital words with respect to amplitudes represented by thereceived digital words.
 70. Decoder according to claim 68, wherein themedia signal additionally comprises a further received stream of digitalwords representing encoded primary media data, the further receivedstream being separate from the received stream, and wherein the decoderis configured to process the primary media data represented by thefurther received stream using the metadata or control data representedby the bitstream.
 71. Decoder according to claim 68 comprising: areference pattern analyzer for analyzing a reference pattern of theencoded secondary media data, wherein the reference pattern analyzer isconfigured to determine an amplitude of the reference pattern or todetermine a predetermined timing instant in the primary media data; asignal manipulator for manipulating the encoded secondary media data inaccordance with the analyzed reference pattern and a computed referencepattern to obtain secondary media data; a signal processor forprocessing the primary media data according to the encoded secondarymedia data to obtain a decoded media signal.
 72. Decoder according toclaim 68, wherein a signal manipulator comprises a sample rate converterconfigured to convert a sample rate associated with the digital words,according to a predetermined timing instant of the primary media dataindicated in the reference pattern, to a predetermined sample rate toobtain resampled digital words.
 73. Decoder according to claim 68,wherein a reference pattern analyzer comprises: a timing instantdeterminer configured to determine the predefined timing instant of theprimary media data in the reference pattern in terms of samples of asample rate; an upsampler configured to upsample a range around thedetermined timing instant to determine an exact position of apredetermined timing instant; a sampling accumulator configured todetermine an exact position of the digital words within the stream ofdigital words to obtain an actual sample rate associated to the digitalwords being different from a predetermined sample rate.
 74. Decoderaccording to claim 68, wherein a reference pattern analyzer comprises again factor calculator to calculate an amplification or attenuationfactor according to the amplitude of the reference pattern and theamplitude of the computed reference pattern and wherein the signalmanipulator comprises a multiplier configured to amplify or attenuatethe data words according to the amplification or attenuation factor toobtain gain compensated data words.
 75. Decoder according to claim 68,wherein the reference pattern analyzer comprises an amplitude detectorto determine an amplitude of a reference pattern and a further amplitudeof the reference pattern; wherein the reference pattern analyzer furthercomprises an offset compensation unit configured to calculate an offsetof the encoded secondary media data according to a drift of theamplitude of the reference pattern and the further amplitude of thereference pattern, wherein the signal manipulator comprises an adderconfigured to add the offset of the encoded secondary media data to theencoded secondary media data to obtain offset compensated encodedsecondary media data.
 76. Decoder according to claim 68, wherein asignal manipulator comprises a demapper configured to demap secondarymedia data comprising a first number of bits from data words comprisinga second number of bits being greater than the first number of bits; orwherein a signal manipulator comprises an ungrouper configured toungroup a group of secondary media data comprising a first number ofbits to obtain the bitstream of decoded secondary media data.
 77. Methodfor decoding a media signal comprising a received stream of digitalwords representing encoded secondary media data comprising metadata orcontrol data for primary media data, the method comprising: recoveringthe secondary media data, wherein the recovering comprises applying adigital demodulation operation or a resampling operation to obtainrecovered secondary media data, and deriving a bitstream from therecovered secondary media data.
 78. Method for encoding secondary mediadata comprising metadata or control data for primary media data, themethod comprising: encoding the secondary media data to obtain a streamof digital words, the encoding comprising transforming the secondarymedia data by a digital modulation or comprising bandlimiting, andoutputting the encoded secondary media data as the stream of digitalwords.
 79. A non-transitory digital storage medium having a computerprogram stored thereon to perform the method for decoding a media signalcomprising a received stream of digital words representing encodedsecondary media data comprising metadata or control data for primarymedia data, the method comprising: recovering the secondary media data,wherein the recovering comprises applying a digital demodulationoperation or a resampling operation to obtain recovered secondary mediadata, and deriving a bitstream from the recovered secondary media data,when said computer program is run by a computer.
 80. A non-transitorydigital storage medium having a computer program stored thereon toperform the method for encoding secondary media data comprising metadataor control data for primary media data, the method comprising: encodingthe secondary media data to obtain a stream of digital words, theencoding comprising transforming the secondary media data by a digitalmodulation or comprising bandlimiting, and outputting the encodedsecondary media data as the stream of digital words, when said computerprogram is run by a computer.
 81. Data processing system comprising: anencoder in accordance of claim 46; and a decoder for decoding a mediasignal comprising a received stream of digital words representingencoded secondary media data comprising metadata or control data forprimary media data; wherein the decoder is configured to recover thesecondary media data, wherein the recovering comprises applying adigital demodulation operation or a resampling operation to obtainrecovered secondary media data, and wherein the decoder is configured toderive a bitstream from the recovered secondary media data.
 82. The dataprocessing system of claim 81, further comprising: a signal manipulatorfor manipulating the stream of digital words to obtain a manipulatedstream of digital words, wherein the decoder is configured to recoverthe stream of digital words from the manipulated stream of digitalwords.
 83. The data processing system of claim 82, wherein the signalmanipulator is configured to manipulate by amplitude amplification oramplitude attenuation or offset introduction or offset variation orfrequency selective attenuation or amplification or resampling, andwherein the decoder is configured to recover the stream of digital wordsmanipulated by amplitude amplification or amplitude attenuation oroffset introduction or offset variation or frequency selectiveattenuation or amplification or resampling.
 84. The data processingsystem of claim 82, wherein the signal manipulator is configured toreceive a PCM audio channel and to output a PCM audio channel, whereinthe encoder is configured to output a signal transmittable over the PCMaudio channel, and wherein the decoder is configured to receive thereceived stream from the PCM audio channel.